-----Original Message----- From: stan [mailto:ghjeold_i_mwee@cox.net] Sent: Tuesday, July 01, 2008 7:20 PM To: Mitul Sen (misen) Cc: alsa-devel@alsa-project.org Subject: Re: [alsa-devel] Setting format to SND_PCM_FORMAT_MU_LAW does not let me apply hardware parameters
Mitul Sen (misen) wrote:
Hi Stan,
Thanks for all your help! I have some more questions though...
I downloaded the source code for alsa-lib-1.0.15 Based on
the code, if
the format is SND_PCM_FORMAT_MU_LAW, I am not sure why it does a get/put index to SND_PCM_FORMAT_S16 Also, if the stream is SND_PCM_STREAM_PLAYBACK, then I would think that it should
decode the
data. Why does it call snd_pcm_mulaw_decode function if the
format is
SND_PCM_FORMAT_MU_LAW and snd_pcm_mulaw_encode otherwise. I have an Intel HDA soundcard and according to the specs, it should
support PCM
ulaw format.
All ALSA documentation and examples I have come across use specific hw_params (like sample rate of 44100, SND_PCM_FORMAT_S16, 2 channel interleaved data). According to the documents, hw_params
refer to the
stream related info so that's the reason I tried to change
it to that
of mu-law (sampling rate of 8000 Hz, SND_PCM_FORMAT_MU_LAW
etc). Not
sure if that's the way to do it though. Based on the code it looks like the hardware just seems to support SND_PCM_FORMAT_S16. Any pointers to help me better understand the ALSA code would
be much appreciated.
Hi Misen,
First, a gentle remonstrance. You probably have noticed that I always put my responses after or mixed with your message. On public mailing lists this is considered good form, rather than posting your response at the top of the message. Why? So that anyone who steps into the interaction doesn't have to read the messages out of order and that future searchers have an easier time understanding the message. While top posting is the norm in communications between two or a few people because the context is familiar to all and it saves time not to have to look for the response, on a public mailing list that isn't necessarily true.
Point noted!
Now to the matter at hand. I had never heard of mu law so I looked it up. http://www.digitalpreservation.gov/formats/fdd/fdd000039.shtml ... Standard companding algorithm used in digital communications systems in North America and Japan (telephones, for the most part) to optimize the dynamic range of an analog signal (generally a voice) for digitizing, i.e., to compress 16 bit LPCM http://www.digitalpreservation.gov/formats/fdd/fdd000011.shtm l (Linear Pulse Code Modulated) data down to 8 bits of logarithmic data. See also Notes http://www.digitalpreservation.gov/formats/fdd/fdd000039.shtml#notes below. ยต-Law is similar to the A-Law http://www.digitalpreservation.gov/formats/fdd/fdd000038.shtml algorithm used in Europe. ...
The code that you extracted below is designed to convert mu law from the compressed form back into the 16 bit signed form. I haven't checked the rest of the code myself, but it appears to assume that the sound device is incapable of internal conversion. If that is true, you shouldn't have to specify anything else to the library except mu law. It should take care of everything else. i.e. as soon as you specify mu law, it is known that the stream is 8 bit mono that has to be uncompressed to 16 bit mono. I presume that is why there is the error when you try to set the hardware parms with mu law.
Thanks! You are right. I don't get the error when I only specify mu law and let the library convert it to 16 bit signed form. I can now hear "some" audio but it sounds very faint. I have been playing around with the frame size because I seem to be able to hear the audio only when I use a small frame size. I need to get a better understanding of the code and my sound device to improve the audio quality. For now, at least I am able to set the hardware parms with mulaw.
The library should probably be modified to use this new capability of sound device internal conversion for mu law if it is available on the sound device. Maybe it already does; as I said I haven't looked at the code, and I'm not really familiar with mu law.
So, given my ignorance, my explanation and proposed solution might be completely wrong. :-) Perhaps a developer familiar with the coding of mu law will give a better explanation.
At this point, I really don't have more to offer for your problem. I would have to look at the code to decipher it in order to give an answer. You might as well do that yourself, as you will get a better understanding than I could give with an explanation.
The code that I am referring to is in pcm_mulaw.c and is as
follows:-
static int snd_pcm_mulaw_hw_params(snd_pcm_t *pcm,
snd_pcm_hw_params_t
params) { snd_pcm_mulaw_t *mulaw = pcm->private_data; snd_pcm_format_t format; int err = snd_pcm_hw_params_slave(pcm, params,
snd_pcm_mulaw_hw_refine_cchange,
snd_pcm_mulaw_hw_refine_sprepare,
snd_pcm_mulaw_hw_refine_schange,
snd_pcm_generic_hw_params);
if (err < 0) return err; err =INTERNAL(snd_pcm_hw_params_get_format)(params, &format);
if (err < 0) return err; if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { if (mulaw->sformat == SND_PCM_FORMAT_MU_LAW) { mulaw->getput_idx =snd_pcm_linear_get_index(format, SND_PCM_FORMAT_S16); mulaw->func = snd_pcm_mulaw_encode; } else { mulaw->getput_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, mulaw->sformat); mulaw->func = snd_pcm_mulaw_decode; } } else { if (mulaw->sformat == SND_PCM_FORMAT_MU_LAW) { mulaw->getput_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, format); mulaw->func = snd_pcm_mulaw_decode; } else { mulaw->getput_idx = snd_pcm_linear_get_index(mulaw->sformat, SND_PCM_FORMAT_S16); mulaw->func = snd_pcm_mulaw_encode; } } return 0; }
Thanks and regards, Mitul
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